- 30 Oct, 2020 11 commits
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Jernej Skrabec authored
H6 I2S is very similar to H3, except that it supports up to 16 channels and thus few registers have fields on different position. Signed-off-by:
Jernej Skrabec <jernej.skrabec@siol.net> Signed-off-by:
Marcus Cooper <codekipper@gmail.com> Acked-by:
Maxime Ripard <mripard@kernel.org> Acked-by:
Rob Herring <robh@kernel.org> Acked-by:
Chen-Yu Tsai <wens@csie.org> Signed-off-by:
Clément Péron <peron.clem@gmail.com> Link: https://lore.kernel.org/r/20201030144648.397824-11-peron.clem@gmail.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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Clément Péron authored
Checkpatch script produces warning: WARNING: function definition argument 'const struct sun4i_i2s *' should also have an identifier name. Let's fix this by adding identifier name to get_bclk_parent_rate() and set_fmt() callback definition. Acked-by:
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Samuel Holland authored
Because SUN4I_I2S_FIFO_CTRL_REG is volatile, writes done while the regmap is cache-only are ignored. To work around this, move the configuration to a callback that runs while the ASoC core has a runtime PM reference to the device. Signed-off-by:
Samuel Holland <samuel@sholland.org> Reviewed-by:
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Clément Péron authored
The FIFO TX reg is volatile and sun8i i2s register mapping is different from sun4i. Even if in this case it's doesn't create an issue, Avoid setting some regs that are undefined in sun8i. Acked-by:
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Marcus Cooper authored
Extend the functionality of the driver to include support of 20 and 24 bits per sample. Signed-off-by:
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Marcus Cooper authored
On the newer SoCs such as the H3 and A64 this is set by default to transfer a 0 after each sample in each slot. However the A10 and A20 SoCs that this driver was developed on had a default setting where it padded the audio gain with zeros. This isn't a problem while we have only support for 16bit audio but with larger sample resolution rates in the pipeline then SEXT bits should be cleared so that they also pad at the LSB. Without this the audio gets distorted. Set sign extend sample for all the sunxi generations even if they are not affected. This will keep consistency and avoid relying on default. Signed-off-by:
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Clément Péron authored
We are actually using a complex formula to just return a bunch of simple values. Also this formula is wrong for sun4i when calling get_wss() the function return 4 instead of 3. Replace this with a simpler switch case. Also drop the i2s params which is unused and return a simple int as returning an error code could be out of range for an s8 and there is no optim to return a s8 here. Fixes: 619c15f7 ("ASoC: sun4i-i2s: Change SR and WSS computation") Reviewed-by:
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Jernej Skrabec authored
H6 I2S is very similar to that in H3, except it supports up to 16 channels. Signed-off-by:
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Clément Péron authored
As slots and slot_width can be set manually using set_tdm(). These values are then kept in sun4i_i2s struct. So we need to check if these values are set or not. This is not done actually and will trigger a bug. For example, if we set to the simple soundcard in the device-tree dai-tdm-slot-width = <32> and then start a stream using S16_LE, currently we would calculate BCLK for 32-bit slots, but program lrck_period for 16-bit slots, making the sample rate double what we expected. To fix this, we need to check if these values are set or not but as this logic is already done by the caller. Avoid duplicating this logic and just pass the required values as params to set_chan_cfg(). Suggested-by:
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Clément Péron authored
Left and Right justified mode are computed using the same formula as DSP_A and DSP_B mode. Which is wrong and the user manual explicitly says: LRCK_PERDIOD: PCM Mode: Number of BCLKs within (Left + Right) channel width. I2S/Left-Justified/Right-Justified Mode: Number of BCLKs within each individual channel width(Left or Right) Fix this by using the same formula as the I2S mode. Fixes: 7ae7834e ("ASoC: sun4i-i2s: Add support for DSP formats") Signed-off-by:
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Michał Mirosław authored
Make tegra20-spdif default to N as all other drivers do. Signed-off-by:
Michał Mirosław <mirq-linux@rere.qmqm.pl> Fixes: 774fec33 ("ASoC: Tegra: Implement SPDIF CPU DAI") Link: https://lore.kernel.org/r/8756eb5aac561173aa222c9cb64dd314ab1b1f9b.1603925200.git.mirq-linux@rere.qmqm.plSigned-off-by:
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- 29 Oct, 2020 2 commits
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Srinivas Kandagatla authored
Update compatible string as board compatible and device compatible should not be same!. Make the driver inline with the new bindings. Fixes: aa2e2785 ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support") Reported-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20201029101550.31695-2-srinivas.kandagatla@linaro.orgSigned-off-by:
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Srinivas Kandagatla authored
Update compatible string as board compatible and device compatible should not be same!. New compatible is now suffixed with -sndcard to be inline with other Qualcomm Sound cards. This also fixes the warnings/error reported by dt_binding_check. Fixes: 765c3759 ("ASoC: qcom: dt-bindings: Add SM8250 sound card bindings") Reported-by:
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- 28 Oct, 2020 2 commits
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Colin Ian King authored
The check for an error return from the call to snd_pcm_format_width is never true as the unsigned int bitwidth can never be less than zero. Fix this by making bitwidth an int. Fixes: 7cb37b7b ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by:
Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20201028115112.109017-1-colin.king@canonical.comSigned-off-by:
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Srinivas Kandagatla authored
Static analysis Coverity had detected a potential array out-of-bounds write issue due to the fact that MAX AFE port Id was set to 16 instead of using AFE_PORT_MAX macro. Fix this by properly using AFE_PORT_MAX macro. Fixes: aa2e2785 ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support") Reported-by:
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- 27 Oct, 2020 4 commits
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Mark Brown authored
Merge series "Add documentation and machine driver for SC7180 sound card" from Cheng-Yi Chiang <cychiang@chromium.org>: Note: - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang@chromium.org> Rohit kumar <rohitkr@codeaurora.org> Ajit Pandey <ajitp@codeaurora.org> But Ajit has left codeaurora. Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi@google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Changes from v5 to v6: - Documentation: Addressed suggestions from Rob. - Drop contains in compatible strings. - Only allow dai-link@[0-9] - Remove reg ref since it has a type definition already. Changes from v6 to v7 - Documentation: - Add headset-jack and hdmi-jack to specify the codec responsible for jack detection. - HDMI codec driver: - Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect. - Machine driver: - Removed aux device following Stephan's suggestion. - Use headset-jack and hdmi-jack to specify the codec responsible for jack detection. - Add support for HDMI(actually DP) playback. Changes from v7 to v8 - Documentation: - Remove headset-jack and hdmi-jack. - Machine driver: - Let machine driver decide whether there is a jack on the DAI. Changes from v8 to v9 - hdmi-codec driver: - Fixed the naming. - Machine driver: - Fixed unused fields. - Moved snd_soc_card_set_drvdata - Keep the naming of HDMI as dai name until v5 of lpass-hdmi patches. Changes from v9 to v10 - Documentation: - Let compatible string be more specific for board configuration to allow for future changes. - Machine driver: - Fixed unused include and macro. - Add temporary macro SC7180_LPASS_DP for future change in sc7180-lpass.h. - Let sound card be dynamically allocated. - Change compatible string accordingly. Changes from v10 to v11 - Machine driver: - Use temporary macro LPASS_DP_RX for future change in sc7180-lpass.h. Changes from v11 to v12 - Documentation: - Change the file and title name for new compatible string google,sc7180-trogdor. - Change the example of model name. - Machine driver: - Use the definitaion of index LPASS_DP_RX in sc7180-lpass.h. - Fix for compatible string. - Replace a comma with semicolon. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (1): ASoC: google: dt-bindings: Add sc7180-trogdor machine bindings .../bindings/sound/google,sc7180-trogdor.yaml | 130 +++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 266 ++++++++++++++++++ 4 files changed, 410 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.29.0.rc2.309.g374f81d7ae-goog -
Srinivas Kandagatla authored
make use of devm_of_platform_populate to remove some redundant code! Signed-off-by:
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Ajit Pandey authored
Add new driver to register sound card on sc7180 trogdor board and do the required configuration for lpass cpu dai and external codecs connected over MI2S interfaces. Signed-off-by:
Ajit Pandey <ajitp@codeaurora.org> Signed-off-by:
Cheng-Yi Chiang <cychiang@chromium.org> Reviewed-by:
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Cheng-Yi Chiang authored
Add devicetree bindings documentation file for sc7180 sound card. Signed-off-by:
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- 26 Oct, 2020 21 commits
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Mark Brown authored
Series with multiple code cleanups, plus one fix to remove unnecessary kernel warnings related to firmware loading. Bard Liao (1): ASoC: SOF: loader: handle all SOF_IPC_EXT types Pierre-Louis Bossart (3): ASoC: SOF: control: remove const in sizeof() ASoC: SOF: topology: remove const in sizeof() ASoC: SOF: sof-audio: remove goto used for force-nocodec support sound/soc/sof/control.c | 20 ++++++++++---------- sound/soc/sof/loader.c | 5 +++++ sound/soc/sof/sof-audio.c | 10 +++------- sound/soc/sof/topology.c | 2 +- 4 files changed, 19 insertions(+), 18 deletions(-) -- 2.27.0
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Mark Brown authored
Viorel Suman <viorel.suman@nxp.com>: From: Viorel Suman <viorel.suman@nxp.com> DAI driver for new XCVR IP found in i.MX8MP. Viorel Suman (2): ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver ASoC: dt-bindings: fsl_xcvr: Add document for XCVR Changes since v1: - improved 6- and 12-ch layout comment - used regmap polling function, improved clocks handling in runtime_resume - added FW size check in FW load function, improved IRQ handler, removed dummy IRQ handlers - fixed yaml file Changes since v2: - used devm_reset_control_get_exclusive instead of of_reset_control_get - moved reset_control_assert into runtime_suspend Changes since v3: - removed "firmware-name" DTS property from both documentation and source code by porting it into SoC specific 'compatible' data structure. .../devicetree/bindings/sound/fsl,xcvr.yaml | 104 ++ sound/soc/fsl/Kconfig | 10 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_xcvr.c | 1359 +++++++++++++++++ sound/soc/fsl/fsl_xcvr.h | 266 ++++ 5 files changed, 1741 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,xcvr.yaml create mode 100644 sound/soc/fsl/fsl_xcvr.c create mode 100644 sound/soc/fsl/fsl_xcvr.h -- 2.26.2
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Mark Brown authored
Merge series "ASoC: merge soc_pcm_hw_param() rollback and soc_pcm_hw_free()" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>: Hi Mark soc_pcm_hw_params() does rollback when failed (A), but, it is almost same as soc_pcm_hw_free(). static int soc_pcm_hw_params(xxx) { ... if (ret < 0) goto xxx_err; ... return ret; ^ component_err: | ... | interface_err: (A) ... | codec_err: | ... v return ret; } This kind of duplicated code can be a hotbed of bugs, thus, this patch-set share soc_pcm_hw_free() and rollback. Kuninori Morimoto (6): ASoC: soc.h: remove for_each_rtd_dais_rollback() ASoC: soc-pcm: move soc_pcm_hw_free() next to soc_pcm_hw_params() ASoC: soc-link: add mark for snd_soc_link_hw_params/free() ASoC: soc-component: add mark for snd_soc_pcm_component_hw_params/free() ASoC: soc-dai: add mark for snd_soc_dai_hw_params/free() ASoC: soc-pcm: add soc_pcm_hw_clean() and call it from soc_pcm_hw_params/free() include/sound/soc-component.h | 6 +- include/sound/soc-dai.h | 4 +- include/sound/soc-link.h | 3 +- include/sound/soc.h | 7 +- sound/soc/soc-component.c | 19 ++--- sound/soc/soc-dai.c | 13 +++- sound/soc/soc-dapm.c | 4 +- sound/soc/soc-link.c | 12 +++- sound/soc/soc-pcm.c | 131 ++++++++++++++-------------------- 9 files changed, 97 insertions(+), 102 deletions(-) -- 2.25.1 -
Mark Brown authored
Merge series "ASoC: qcom: add support for QRB5165 RB5 machine" from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>: This patchset adds support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. Srinivas Kandagatla (2): ASoC: qcom: dt-bindings: Add SM8250 sound card bindings ASoC: qcom: sm8250: add sound card qrb5165-rb5 support .../bindings/sound/qcom,sm8250.yaml | 161 +++++++++++++ sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sm8250.c | 228 ++++++++++++++++++ 4 files changed, 402 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sm8250.yaml create mode 100644 sound/soc/qcom/sm8250.c -- 2.21.0
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Mark Brown authored
Merge series "use semicolons rather than commas to separate statements" from Julia Lawall <Julia.Lawall@inria.fr>: These patches replace commas by semicolons. This was done using the Coccinelle semantic patch (http://coccinelle.lip6.fr/) shown below. This semantic patch ensures that commas inside for loop headers will not be transformed. It also doesn't touch macro definitions. Coccinelle ensures that braces are added as needed when a single-statement branch turns into a multi-statement one. This semantic patch has a few false positives, for variable delcarations such as: LIST_HEAD(x), *y; The semantic patch could be improved to avoid these, but for the moment they have been removed manually (2 occurrences). // <smpl> @initialize:ocaml@ @@ let infunction p = (* avoid macros *) (List.hd p).current_element <> "something_else" let combined p1 p2 = (List.hd p1).line_end = (List.hd p2).line || (((List.hd p1).line_end < (List.hd p2).line) && ((List.hd p1).col < (List.hd p2).col)) @bad@ statement S; declaration d; position p; @@ S@p d // special cases where newlines are needed (hope for no more than 5) @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @r@ expression e1,e2; statement S; position p != bad.p; @@ e1 ,@S@p e2; @@ expression e1,e2; position p1; position p2 : script:ocaml(p1) { infunction p1 && not(combined p1 p2) }; statement S; position r.p; @@ e1@p1 -,@S@p +; e2@p2 ... when any // </smpl> --- sound/firewire/fireworks/fireworks_pcm.c | 2 +- sound/pci/hda/patch_ca0132.c | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/soc/codecs/madera.c | 4 ++-- sound/soc/codecs/wm8350.c | 3 ++- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/soc-dapm.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 2 +- 9 files changed, 11 insertions(+), 10 deletions(-)
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Mark Brown authored
Merge series "ASoC: sun8i-codec: support for AIF2 and AIF3" from Samuel Holland <samuel@sholland.org>: This series adds support the other two AIFs present in the sun8i codec, which can be used for codec2codec DAI links. This series first fills out the DAI driver, removing assumptions that were made for AIF1 (16 bits, 2 channels, certain clock inversions). Some new logic is required to handle 3 DAIs and the ADC/DAC sharing the same clock. Finally, it adds the new DAIs, and hooks them up with DAPM widgets and routes per the hardware topology. To minimize the number of patches in this series, related device tree patches (increasing #sound-dai-cells, adding new DAI links) will be sent separately. Changes from v1: - Patches 1-8 from v1 (DAPM changes) were merged - Prefixed AIF constants with "SUN8I_CODEC_" [1, 7, 10, 16, 17] - Renamed variables in sun8i_codec_set_fmt for clarity [3] - Update sysclk->sysclk_rate if later calls to hw_params change the sample rate (thanks Chen-Yu for reminding me of this) [11] - Select COMMON_CLK for clk_set_rate_exclusive [12] - Add comments and hopefully clarify the clock protection logic [12] - Make the error message more concise and put it on one line [12] - Drop the "reg" variable holding SUN8I_AIF_CLK_CTRL(dai->id) [15] - Rename "div_reg" to "clk_reg" and adjust comments for clarity [17] - Improve the AIF2/AIF3 rate mismatch error message [17] - Minor grammar/wording fixes in commit messages [2, 4, 7, 8, 16, 17] - Added Maxime's Acked-by: [1-2, 4-9, 11, 13-14, 16] Samuel Holland (17): ASoC: sun8i-codec: Prepare to extend the DAI driver ASoC: sun8i-codec: Program DAI format before clock inversion ASoC: sun8i-codec: Enable all supported clock inversions ASoC: sun8i-codec: Use the provided word size ASoC: sun8i-codec: Round up the LRCK divisor ASoC: sun8i-codec: Correct the BCLK divisor calculation ASoC: sun8i-codec: Support the TDM slot binding ASoC: sun8i-codec: Enforce symmetric DAI parameters ASoC: sun8i-codec: Enable all supported sample rates ASoC: sun8i-codec: Automatically set the system sample rate ASoC: sun8i-codec: Constrain to compatible sample rates ASoC: sun8i-codec: Protect the clock rate while streams are open ASoC: sun8i-codec: Require an exact BCLK divisor match ASoC: sun8i-codec: Enable all supported PCM formats ASoC: sun8i-codec: Generalize AIF clock control ASoC: sun8i-codec: Add the AIF2 DAI, widgets, and routes ASoC: sun8i-codec: Add the AIF3 DAI, widgets, and routes sound/soc/sunxi/Kconfig | 1 + sound/soc/sunxi/sun8i-codec.c | 834 ++++++++++++++++++++++++++++------ 2 files changed, 699 insertions(+), 136 deletions(-) -- 2.26.2 -
Mark Brown authored
Merge series "dt-bindings: stm32: convert audio dfsdm to json-schema" from Olivier Moysan <olivier.moysan@st.com>: Some audio properties documented in st,stm32-adfsdm.txt are already documented in Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml bindings. Move remaining properties from st,stm32-adfsdm.txt to st,stm32-dfsdm-adc.yaml, and remove st,stm32-adfsdm.txt. Changes in v2: - Complete st,stm32-dfsdm-adc.yaml rather than converting st,stm32-adfsdm.txt Olivier Moysan (2): dt-bindings: stm32: dfsdm: update audio properties dt-bindings: stm32: dfsdm: remove stm32-adfsdm.txt binding .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 7 ++- .../bindings/sound/st,stm32-adfsdm.txt | 63 ------------------- 2 files changed, 5 insertions(+), 65 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt -- 2.17.1
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Shengjiu Wang authored
Add 88200Hz and 176400Hz sample rates support for TX. Add 88200Hz, 176400Hz, 192000Hz sample rates support for RX. Signed-off-by:
Viorel Suman <viorel.suman@nxp.com> Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by:
Nicolin Chen <nicoleotsuka@gmail.com> Reviewed-by:
Daniel Baluta <daniel.baluta@nxp.com> Link: https://lore.kernel.org/r/1602557360-18795-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by:
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Shengjiu Wang authored
On i.MX8QM, there are separate interrupts for TX and RX. As the EDMA can't be configured to swing back to first FIFO after writing the second FIFO, so we need to force the burst size to be 2 on i.MX8QM. And EDMA don't support to shift the data from S24_LE to S16_LE, so the supported TX format is also different on i.MX8QM. Signed-off-by:
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Shengjiu Wang authored
Add new compatible string "fsl,imx8qm-spdif" for supporting spdif module on i.MX8QM. Signed-off-by:
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Vijendar Mukunda authored
ACP supports different audio configurations other than I2S. This patch will fix acp driver probe failure for other audio configurations. Tested-by:
Prike Liang <Prike.Liang@amd.com> Reviewed-by:
Alex Deucher <alexander.deucher@amd.com> Signed-off-by:
Vijendar Mukunda <Vijendar.Mukunda@amd.com> Link: https://lore.kernel.org/r/1603476441-3506-1-git-send-email-Vijendar.Mukunda@amd.comSigned-off-by:
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Richard Fitzgerald authored
Replace the two-step copy-and-convert in wm5102_out_comp_coeff_put() with get_unaligned_be16(). Apart from looking nicer, it avoids this sparse warning: wm5102.c:687:35: sparse: sparse: cast to restricted __be16 Signed-off-by:
Richard Fitzgerald <rf@opensource.cirrus.com> Link: https://lore.kernel.org/r/20201015102703.24622-1-rf@opensource.cirrus.comSigned-off-by:
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Olivier Moysan authored
Convert the STM32 SAI bindings to DT schema format using json-schema. Signed-off-by:
Olivier Moysan <olivier.moysan@st.com> Reviewed-by:
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Alexandru Ardelean authored
The change removes the platform_data include/definition. It only contains some values for the MICBIAS. These are moved into 'dt-bindings/sound/adi,adau1977.h' so that they can be used inside device-trees. When moving then, they need to be converted to pre-compiler defines, so that the DT compiler can understand them. The driver then, also needs to include the new 'dt-bindings/sound/adi,adau1977.h' file. Signed-off-by:
Alexandru Ardelean <alexandru.ardelean@analog.com> Link: https://lore.kernel.org/r/20201019105313.24862-1-alexandru.ardelean@analog.comSigned-off-by:
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Codrin Ciubotariu authored
The TI PCM5102A codec driver can be used with the generic sound card drivers, so it should be selectable. For example, with the addition of #sound-dai-cells = <0> property in DT, it can be used with simple/graph card drivers. Signed-off-by:
Codrin Ciubotariu <codrin.ciubotariu@microchip.com> Link: https://lore.kernel.org/r/20201012141911.3150996-1-codrin.ciubotariu@microchip.comSigned-off-by:
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Lars-Peter Clausen authored
All platforms that use the jz4740-i2s driver have been switched to devicetree for a while now and the assignment of the DMA peripheral ID is done in the devicetree. It is no longer necessary to manually assign the peripheral ID in the driver, so remove that. The DMA driver does not even look at the value assigned in the driver anymore and always uses the value provided by the devicetree. Signed-off-by:
Lars-Peter Clausen <lars@metafoo.de> Reviewed-by:
Paul Cercueil <paul@crapouillou.net> Link: https://lore.kernel.org/r/20201023092346.5777-1-lars@metafoo.deSigned-off-by:
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Srinivas Kandagatla authored
Add support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. Signed-off-by:
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Srinivas Kandagatla authored
This patch adds bindings required for SM8250 based soundcards for example Qualcomm Robotics RB5 Development Kit which makes use of ADSP and Internal LPASS codec. Signed-off-by:
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Olivier Moysan authored
Device tree audio configuration for STM32 DFSDM is already covered in the following binding: Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml Remove stm32-adfsdm.txt obsolete binding. Signed-off-by:
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Olivier Moysan authored
- Add missing compatible property in audio node. - Remove obsolete "st,stm32-dfsdm-pdm" compatible. - Remove useless comment in adc io-channels description. Signed-off-by:
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Kuninori Morimoto authored
soc_pcm_hw_params() does rollback when failed (A), but, it is almost same as soc_pcm_hw_free(). static int soc_pcm_hw_params(xxx) { ... if (ret < 0) goto xxx_err; ... return ret; ^ component_err: | ... | interface_err: (A) ... | codec_err: | ... v return ret; } The difference is soc_pcm_hw_free() is for all dai/component/substream, rollback is for succeeded part only. This kind of duplicated code can be a hotbed of bugs, thus, we want to share soc_pcm_hw_free() and rollback. Now, soc_pcm_hw_params/free() are handling 1) snd_soc_link_hw_params/free() 2) snd_soc_pcm_component_hw_params/free() 3) snd_soc_dai_hw_params/free() Now, 1) to 3) are handled. This patch adds new soc_pcm_hw_clean() and call it from soc_pcm_hw_params() as rollback, and from soc_pcm_hw_free() as normal close handler. Other difference is that soc_pcm_hw_free() handles digital mute if it was last user. Rollback also handles it by this patch. Signed-off-by:Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87h7rhgqab.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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